2 * Copyright (c) 2013 Paul B Mahol
4 * This file is part of FFmpeg.
6 * FFmpeg is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
11 * FFmpeg is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with FFmpeg; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
26 #include "libavutil/avassert.h"
27 #include "libavutil/opt.h"
38 typedef struct AudioPhaserContext {
40 double in_gain, out_gain;
47 int delay_buffer_length;
50 int modulation_buffer_length;
51 int32_t *modulation_buffer;
53 int delay_pos, modulation_pos;
55 void (*phaser)(struct AudioPhaserContext *p,
56 uint8_t * const *src, uint8_t **dst,
57 int nb_samples, int channels);
60 #define OFFSET(x) offsetof(AudioPhaserContext, x)
61 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
63 static const AVOption aphaser_options[] = {
64 { "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, 1, FLAGS },
65 { "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0, 1e9, FLAGS },
66 { "delay", "set delay in milliseconds", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=3.}, 0, 5, FLAGS },
67 { "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, .99, FLAGS },
68 { "speed", "set modulation speed", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1, 2, FLAGS },
69 { "type", "set modulation type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, "type" },
70 { "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
71 { "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
72 { "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
73 { "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
77 AVFILTER_DEFINE_CLASS(aphaser);
79 static av_cold int init(AVFilterContext *ctx)
81 AudioPhaserContext *p = ctx->priv;
83 if (p->in_gain > (1 - p->decay * p->decay))
84 av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
85 if (p->in_gain / (1 - p->decay) > 1 / p->out_gain)
86 av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");
91 static int query_formats(AVFilterContext *ctx)
93 AVFilterFormats *formats;
94 AVFilterChannelLayouts *layouts;
95 static const enum AVSampleFormat sample_fmts[] = {
96 AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
97 AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
98 AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
99 AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
103 layouts = ff_all_channel_layouts();
105 return AVERROR(ENOMEM);
106 ff_set_common_channel_layouts(ctx, layouts);
108 formats = ff_make_format_list(sample_fmts);
110 return AVERROR(ENOMEM);
111 ff_set_common_formats(ctx, formats);
113 formats = ff_all_samplerates();
115 return AVERROR(ENOMEM);
116 ff_set_common_samplerates(ctx, formats);
121 static void generate_wave_table(enum WaveType wave_type, enum AVSampleFormat sample_fmt,
122 void *table, int table_size,
123 double min, double max, double phase)
125 uint32_t i, phase_offset = phase / M_PI / 2 * table_size + 0.5;
127 for (i = 0; i < table_size; i++) {
128 uint32_t point = (i + phase_offset) % table_size;
133 d = (sin((double)point / table_size * 2 * M_PI) + 1) / 2;
136 d = (double)point * 2 / table_size;
137 switch (4 * point / table_size) {
138 case 0: d = d + 0.5; break;
140 case 2: d = 1.5 - d; break;
141 case 3: d = d - 1.5; break;
148 d = d * (max - min) + min;
149 switch (sample_fmt) {
150 case AV_SAMPLE_FMT_FLT: {
151 float *fp = (float *)table;
155 case AV_SAMPLE_FMT_DBL: {
156 double *dp = (double *)table;
162 d += d < 0 ? -0.5 : 0.5;
163 switch (sample_fmt) {
164 case AV_SAMPLE_FMT_S16: {
169 case AV_SAMPLE_FMT_S32: {
180 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
182 #define PHASER_PLANAR(name, type) \
183 static void phaser_## name ##p(AudioPhaserContext *p, \
184 uint8_t * const *src, uint8_t **dst, \
185 int nb_samples, int channels) \
187 int i, c, delay_pos, modulation_pos; \
189 av_assert0(channels > 0); \
190 for (c = 0; c < channels; c++) { \
191 type *s = (type *)src[c]; \
192 type *d = (type *)dst[c]; \
193 double *buffer = p->delay_buffer + \
194 c * p->delay_buffer_length; \
196 delay_pos = p->delay_pos; \
197 modulation_pos = p->modulation_pos; \
199 for (i = 0; i < nb_samples; i++, s++, d++) { \
200 double v = *s * p->in_gain + buffer[ \
201 MOD(delay_pos + p->modulation_buffer[ \
203 p->delay_buffer_length)] * p->decay; \
205 modulation_pos = MOD(modulation_pos + 1, \
206 p->modulation_buffer_length); \
207 delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \
208 buffer[delay_pos] = v; \
210 *d = v * p->out_gain; \
214 p->delay_pos = delay_pos; \
215 p->modulation_pos = modulation_pos; \
218 #define PHASER(name, type) \
219 static void phaser_## name (AudioPhaserContext *p, \
220 uint8_t * const *src, uint8_t **dst, \
221 int nb_samples, int channels) \
223 int i, c, delay_pos, modulation_pos; \
224 type *s = (type *)src[0]; \
225 type *d = (type *)dst[0]; \
226 double *buffer = p->delay_buffer; \
228 delay_pos = p->delay_pos; \
229 modulation_pos = p->modulation_pos; \
231 for (i = 0; i < nb_samples; i++) { \
232 int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \
233 p->delay_buffer_length) * channels; \
236 delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \
237 npos = delay_pos * channels; \
238 for (c = 0; c < channels; c++, s++, d++) { \
239 double v = *s * p->in_gain + buffer[pos + c] * p->decay; \
241 buffer[npos + c] = v; \
243 *d = v * p->out_gain; \
246 modulation_pos = MOD(modulation_pos + 1, \
247 p->modulation_buffer_length); \
250 p->delay_pos = delay_pos; \
251 p->modulation_pos = modulation_pos; \
254 PHASER_PLANAR(dbl, double)
255 PHASER_PLANAR(flt, float)
256 PHASER_PLANAR(s16, int16_t)
257 PHASER_PLANAR(s32, int32_t)
264 static int config_output(AVFilterLink *outlink)
266 AudioPhaserContext *p = outlink->src->priv;
267 AVFilterLink *inlink = outlink->src->inputs[0];
269 p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5;
270 p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels);
271 p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;
272 p->modulation_buffer = av_malloc(p->modulation_buffer_length * sizeof(*p->modulation_buffer));
274 if (!p->modulation_buffer || !p->delay_buffer)
275 return AVERROR(ENOMEM);
277 generate_wave_table(p->type, AV_SAMPLE_FMT_S32,
278 p->modulation_buffer, p->modulation_buffer_length,
279 1., p->delay_buffer_length, M_PI / 2.0);
281 p->delay_pos = p->modulation_pos = 0;
283 switch (inlink->format) {
284 case AV_SAMPLE_FMT_DBL: p->phaser = phaser_dbl; break;
285 case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break;
286 case AV_SAMPLE_FMT_FLT: p->phaser = phaser_flt; break;
287 case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break;
288 case AV_SAMPLE_FMT_S16: p->phaser = phaser_s16; break;
289 case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break;
290 case AV_SAMPLE_FMT_S32: p->phaser = phaser_s32; break;
291 case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break;
292 default: av_assert0(0);
298 static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
300 AudioPhaserContext *p = inlink->dst->priv;
301 AVFilterLink *outlink = inlink->dst->outputs[0];
304 if (av_frame_is_writable(inbuf)) {
307 outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples);
309 return AVERROR(ENOMEM);
310 av_frame_copy_props(outbuf, inbuf);
313 p->phaser(p, inbuf->extended_data, outbuf->extended_data,
314 outbuf->nb_samples, av_frame_get_channels(outbuf));
317 av_frame_free(&inbuf);
319 return ff_filter_frame(outlink, outbuf);
322 static av_cold void uninit(AVFilterContext *ctx)
324 AudioPhaserContext *p = ctx->priv;
326 av_freep(&p->delay_buffer);
327 av_freep(&p->modulation_buffer);
330 static const AVFilterPad aphaser_inputs[] = {
333 .type = AVMEDIA_TYPE_AUDIO,
334 .filter_frame = filter_frame,
339 static const AVFilterPad aphaser_outputs[] = {
342 .type = AVMEDIA_TYPE_AUDIO,
343 .config_props = config_output,
348 AVFilter avfilter_af_aphaser = {
350 .description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
351 .query_formats = query_formats,
352 .priv_size = sizeof(AudioPhaserContext),
355 .inputs = aphaser_inputs,
356 .outputs = aphaser_outputs,
357 .priv_class = &aphaser_class,